In the communication age, bandwidth is money. Video and audio signals (hereinafter “media signals”) consume enormous amounts of bandwidth depending on the desired transmission quality. As a result, data compression is playing an increasingly important role in communication.
Conventionally, the parties to a communication decide on a particular codec (compressor/decompressor) for compressing and decompressing media signals. A wide variety of codecs are available. General classifications of codecs include discrete cosine transfer (DCT) or “block” codecs, fractal codecs, and wavelet codecs.
Some codecs are “lossless,” meaning that no data is lost during the compression process. A compressed media signal, after being received and decompressed by a lossless codec, is identical to the original. However, most commercially-available codecs are “lossy” and result in some degradation of the original media signal.
For lossy codecs, compression “quality” (i.e., how similar a compressed media signal is to the original after decompression) varies substantially from codec to codec, and may depend, for instance, on the amount of available bandwidth, the quality of the communication line, characteristics of the media signal, etc. Another compression metric, i.e., performance, relates to the amount of bandwidth required to transmit the compressed signal as opposed to the original signal. Typically, lossy codecs result in better performance than lossless codecs, which is why they are preferred in most applications.
Codec designers generally attempt to fashion codecs that produce high quality compressed output across a wide range of operating parameters. Although some codecs, such as MPEG-2, have gained widespread acceptance because of their general usefulness, no codec is ideally suited to all purposes. Each codec has individual strengths and weaknesses.
Conventionally, the same codec is used to compress and decompress a media signal during the entire communication session or uniformly across a storage medium (e.g., DVD). However, a media signal is not a static quantity. A video signal, for example, may change substantially from scene to scene. Likewise, the available bandwidth or line quality may change during the course of a communication. Selecting the wrong codec at the outset can be a costly mistake in terms of the bandwidth required to transmit or store the media signal.
Another problem arises from the selection of various codec settings, which typically apply throughout the communication session. Because the codec settings affect the “quality” of the transmission, i.e., how similar a received and decompressed signal is to the original, such settings are often referred to as quality settings.
In general, quality settings affect the amount of bandwidth required for the transmission. Higher quality settings typically consume greater bandwidth, while lower quality settings require lesser bandwidth.
Unfortunately, the bandwidth required for sending each frame of a media signal is variable, as is the overall amount of available bandwidth. Using a single set of quality settings throughout a transmission does not take into account this variability, and the result is video “jerkiness” (frame loss), audio degradation, and the like, when there is insufficient bandwidth to represent a frame at a given moment in time. Anyone who has participated in a videoconferencing session has experienced the uneven quality of conventional approaches.